2nd call not picking up/calls out with wrong CID

Discussion in '3CX Phone System - General' started by RLester, Feb 16, 2012.

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  1. RLester

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    I have been evaluating 3CX and like it but I have a strange problem and can’t seem to work it out on my own. I have plans to purchase the 16call version but need to get my troubles worked out first.

    I have 3cx v10 SP6 (had trouble in SP5 as well) Grand Stream GXW4108 up to date FW and (2) Grand Stream GXP2110 phones up to date FW as well.

    This is the trouble…

    GXW4108 with (2) lines hooked to FXO ports 1 & 2. Can call out on both ch1 & ch2 (2 active calls), can call in by dialing each phone number separately hanging up between each (only 1 active call at a time). While ch1 has active call ch2 will not ring internal. While ch2 has active call ch1 will ring internal and can be answered (See my work around description later).

    One thing I have seen is that while looking at 3CX Ports/Trunks Status when making outbound call ch1 shows active call but on the receiving phone it shows ch2 phone number (CID). I have made sure of the line hooked in to ch1 and ch2. In other words I know that ch1 has XXX-XXX-2304 number and ch2 has XXX-XXX-4510 number hooked in but receiving phone always show XXX-XXX-4510 as the caller ID even though 3CX shows ch1 active which has XXX-XXX-2304. Just to be clear, I can call in to XXX-XXX-2304 and 3CX shows active call on ch1 then can call XXX-XXX-4510 and 3CX will show active call on ch2 which is correct but going out it is not. It is only correct if ch2 gets call 1st (See my work around description later).

    Opened support ticket with Grand Stream and they say its 3CX trouble and they suggest this…

    This should solve the problem. In your sip server, set up Prefix on extensions accordingly like this:
    1. Extension registered for FXO port 1 of GXW410x , set up the Prefix(on 3CX Trunk) to 991
    2. Extension registered for FXO port 2 of GXW410x , set up the Prefix(on 3CX Trunk) to 992
    3. Extension registered for FXO port x of GXW410x , set up the Prefix(on 3CX Trunk) to 99x
    This should solve the problem. Thanks!

    I just can’t figurer out how to do what they say will fix my trouble. I have as a work around even before talking to them set up 2 gateways 1 for line 1, 1 for line 2 and in the outbound rules set line 2 to use 1st and line 1 to use 2nd for each rule. I just do not see this working for all 8 lines.

    I would post logs but did not think it would help.
     
  2. jpillow

    jpillow Well-Known Member

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    Have you checked under PSTN devices-->edit ports and assure that the maximum simultaneously calls is set to 1? As far as the caller ID that sounds like an issue you may want to discuss with your Telco. SIP trunks you can change outgoing caller ID in 3cx management console but with analog lines I'm pretty sure caller ID broadcast whatever number is associated with that line.
     
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  3. jpillow

    jpillow Well-Known Member

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    Have you checked under PSTN devices-->edit ports and assure that the maximum simultaneously calls is set to 1? As far as the caller ID that sounds like an issue you may want to discuss with your Telco. SIP trunks you can change outgoing caller ID in 3cx management console but with analog lines I'm pretty sure caller ID broadcast whatever number is associated with that line.
     
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  4. leejor

    leejor Well-Known Member

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    There are two general "camps' of PSTN gateways. Some address individual line selection by having each line use a different port number, some use a prefix in the dial string. It sounds like yours is the latter.
    What Grand stream is telling you is that you need to send a prefix along with the number you are dialling to select the outgoing line. So to go out on line one you could have outbound rules set up so that you would dial 94155551212, in this case the 9 means you want to use the first outgoing trunk in the gateway. call the rule 9 plus (or what have you).
    Begins with 9, 11 digits long, strip 1 (the 9), prefix with 991, send to the gateway trunks.

    This would send 9914155551212 to the gateway, it would see the 991 and know to send the call out on the first line.

    To send a call to the second line, create a rule...8 plus...begins with 8, 11 digits long, strip 1, prefix 992. This would send 9924155551212, and so on.

    Of course you will have to create rules that cover the PSTN (local and long distance) plan in your area and you don't have to use access digits, you can base the routing on areacodes or if it is a 1+ (long distance call).
    3Cx is pretty flexible.
    Of course, you will send caller ID based on which line you call out on unless you subscribe to a hunt group with a common billing number, in that case your one number would be sent out on calls placed from either line. This is something you would have to have arranged with your PSTN provider.
     
  5. RLester

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    OK, after looking at all possibilities that I could live with I see by creating an outbound rule that would prepend 992 on route 1 then prepend 991 on route 2 then the user will not have to dial any extra digits.

    Does this seem to be an acceptable workaround?

    Also is there any one out there using a Grand Stream GXW4108 and if so how do you have it configured as it would relate to VoIP to PSTN calls to auto or just grab the next available line. How do you have your Unconditional Call Forward? This is the only way I can get the gateway to answer on both line (ch1:10000;ch2:10001;ch3-8:;).
     
  6. RLester

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    After one more day of working on this and resetting the gateway this last time flowing the guide as before but most importantly this time knowing exactly what I have to change for DTMF and line volume and what not. I think just not having to save and restart it so much that the gateway does work as configured by the on line guide. I did find that you can not create the gateway and rule with the 3cx wizard and then edit the prepend fields for each route. I mean to say that you can change them around but can not expect them to work correctly. What I have seen happing is…

    Create the gateway and rule with the 3cx wizard and put the 992 in route 1, put 991 in route 2 then click finish. This way calls will go out ch2 1st then ch1 2nd and CID on receiving phone is correct in all cases. You can also do this ascending 991,992 and it works fine.

    Create the gateway and rule with the 3cx wizard and click finish. Then go add the 992 and 991 and the call will go out ch1 but will show ch2 CID on receiving phone. This way the system will not ring for 2nd call but you can call out and make that 2nd active call.

    So it appears to me that this is an 8 line FXO gateway and by way of the guide can be configured for 1 stage dialing but will not work for more than 3 lines (3 routs per rule) because apparently you can not pass through a rule to the next if no route is open. What I want to do is lets say I had 6 lines and rule 1 had route 1 set for 991 route 2 992 route 3 993 then in the list of rules have rule 2 and in it route 1 set for 994 route 2 995 route 3 996. With this in place ones the calls run through rule 1 and route 3 then I get server error and call is dropped. This can be made to work by putting lets say (2) in the box on rule 2 for “begins with”, strip 1 digit and putting rule 2 before rule 1. This way the user has to dial 2 and the number to use rule 2 for line 4-6 and dial just number for Line 1-3 on rule 1. Of cores this is crazy!!!

    I do not want to use 2 stages dialing because the user has no way to know what lines are in uses visually. To me dialing 1XXX-XXXX or 2XXX-XXXX and dialing 1 wait for ton then XXX-XXXX or 2 wait for ton then XXX-XXXX is all the same and this is what everyone would have to do without visually seeing what lines are in uses. If I am missing some how to show on the phone which line not just what phone extension (BLF) is in uses please tell me.

    My point to all this is I can’t uses all 8 lines with out 2 stage dialing or something like 2 stage, and I can’t have all my users spending 30 or 40 seconds to find a line to call out on every time all day long. The documentation on 3CX web site where it says this gateway is supported with right FW and no where does it says to use 2 stage dialing in fact it said to use 1 stage. If it would have said to be fully supported and work you have to uses 2 stage dialing I would not have perched this gateway. If 3CX would be so kind to modify the rules for the Grand Stream gateways to make routs for any or all lines attached then it would work fully with 1 stage dialing.

    Thanks for reading this far I just hate seeing uncompleted posts when I’m looking you know you just never know if someone got it to work or just got rid of it.
     
  7. leejor

    leejor Well-Known Member

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    You seem to be trying to use the gateway in a bit of an unorthodox manner. Most users have only a few outgoing routes, perhaps one for local, one for long distance, ad maybe one in another location. The main group may consist of several trunks but a call about on any one of them can show the caller the same number, this is hunt group option provided by your PSTN provider. For example, four trunks in the group, all showing the "pilot" number. If the pilot number is called back, it will ring on the first available line in the hunt group. At the moment you can only have 3 routes in the outbound rules, but you can use more than one rule, and each can have 3 different trunks. You just have to decide how you want to decide the calls. this can be done by groups of extensions or the numbers dialled.

    It is going to be a lot of work trying to come up with a working system where you are selecting up to 8 outgoing lines and not running into conflicts (hitting a busy trunk). Finding a free trunks should be done by the outbound rules, or the gateway (in the case of one trunk group with many members).
     
  8. RLester

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    I must be misunderstood. All I want to do is plug in 8 lines in the 8 FXO ports and setup one gateway and one rule in 3CX with the wizard. Use 1 stage dialing as the guide says to do and use round robin line selection 1-8. With this setup no one would need to know what line was in uses until you ran out of free lines. I do not want to do all this line selection I just can’t get it to work with out doing this.

    Let me ask is it not reasonable to want to pickup the phone and have 8 lines available and let the system just find the next available line for me to dial my number as XXX-XXXX. This hunt group that you’re talking about is exactly what we have 8 lines, and from the outside world they just roll over to the next available when a call comes in to a number that is in uses. All I want is to do the same thing from inside when I dial a number just find me a line out of the 8 that is not in uses.

    Is this really not possible?
     
  9. lneblett

    lneblett Well-Known Member

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    Ok, let's see if we can back up a little and get some more information. You have an 8 port FXO gateway. Currently, you are using 2 ports for testing. If you look at the PSTN tab on the 3CX system, you should see whatever name you input for the PSTN device (GX4108 for instance). Under that heading should be listed the two ports you created to accommodate the two PSTN lines. As you install more lines to the device, you will also need to install (inform) 3CX of the additioanal lines.

    The first question is related to the inbound, and has to do with how the 2 PSTN lines are configured with your carrier. Do you have roll over or busy call forward implemented? This is usually implemented such that if line one is busy with a call, a caller who used the same number to dial will be rolled to line 2 by the carrier automatically. Also, keep in mind that your caller-id maybe set by your carrier, so what ever line the device elects to call out, the caller-id may be that which the carrier has associated to that particular PSTN line.

    Now then, looking back at the PSTN device tab and the device name, how many simultaneous calls do you have set for the device and then look at each port and how many do you have set there as well? Try setting all to 2 for the moment and try your scenario again and let us know.
     
  10. RLester

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    First off I thank you up front for any help you may provide.

    Rollover is what we call it I do not know is busy call forward the same thing. I see that 1 simultaneous call is set for each account. The wizard sat the simultaneous calls for the gateway I guess; I do not see a setting for the gateway all by itself just for each account.

    If I setup the gateway with prepend numbers with the wizard either like 991,992 or like 992,991. I have no trouble calling out and CID will always be right and the other line is free for in and out calls. No trouble here if all I had was 2 lines to think about.

    What the trouble has been and it was not easy to get to this conclusion but when you do not add the prefix or prepend numbers in the wizard or go back and change them around they will not work correctly. It is like 3cx sends it out on ch1 but the gateway is using ch2 I figure this is why CID of ch2 is sent out and I can not call in but, can call out by way of ch2 in 3cx this time the gateway finds that ch1 is open and will send the call out with CID of ch1. So the port mapping does seem to be a must do in the wizard it’s just I see with out doing something special I could never uses more than 3 lines.

    I will try changing the number of calls on each account to 2 and let you know. This may be about the only thing that I have not tried. Thanks again.
     
  11. RLester

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    Log files show call 62 & 63 I see same thing as before changing the number of concurrent calls.

    what I did is called out 3cx is showing line 1 in use, then trying to call in on line 2 but never rings internal.


    15:28:03.059 [CM503008]: Call(63): Call is terminated
    15:27:55.200 Currently active calls - 1: [63]
    15:27:40.778 [CM503007]: Call(63): Device joined: sip:10010@192.168.0.180:5060;transport=udp
    15:27:40.778 [CM503007]: Call(63): Device joined: sip:400@127.0.0.1:1566;rinstance=ed1e4d4bb16f424b
    15:27:40.763 [CM505002]: Gateway:[PSTNGateway01] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4108 (HW 2.0, Ch:8) 1.3.4.10] PBX contact: [sip:10010@192.168.0.90:5060]
    15:27:40.763 [CM503002]: Call(63): Alerting sip:10010@192.168.0.180:5060;transport=udp
    15:27:36.559 [CM503025]: Call(63): Calling PSTNline:9922651411@(Ln.10010@PSTNGateway01)@[Dev:sip:10010@192.168.0.180:5060;transport=udp]
    15:27:36.513 [CM503004]: Call(63): Route 2: PSTNline:9912651411@(Ln.10010@PSTNGateway01)@[Dev:sip:10010@192.168.0.180:5060;transport=udp,Dev:sip:10020@192.168.0.180:5062;transport=udp]
    15:27:36.513 [CM503004]: Call(63): Route 1: PSTNline:9922651411@(Ln.10010@PSTNGateway01)@[Dev:sip:10010@192.168.0.180:5060;transport=udp,Dev:sip:10020@192.168.0.180:5062;transport=udp]
    15:27:36.513 [CM503010]: Making route(s) to <sip:2651411@192.168.0.90:5060>
    15:27:36.513 [CM505001]: Ext.400: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 6.0.20943.0] PBX contact: [sip:400@127.0.0.1:5060]
    15:27:36.497 [CM503001]: Call(63): Incoming call from Ext.400 to <sip:2651411@192.168.0.90:5060>
    15:26:34.327 [CM503008]: Call(62): Call is terminated
    15:26:19.202 Currently active calls - 1: [62]
    15:25:57.844 [CM503007]: Call(62): Device joined: sip:10010@192.168.0.180:5060;transport=udp
    15:25:57.844 [CM503007]: Call(62): Device joined: sip:400@127.0.0.1:1566;rinstance=ed1e4d4bb16f424b
    15:25:57.781 [CM505002]: Gateway:[PSTNGateway01] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4108 (HW 2.0, Ch:8) 1.3.4.10] PBX contact: [sip:10010@192.168.0.90:5060]
    15:25:57.781 [CM503002]: Call(62): Alerting sip:10010@192.168.0.180:5060;transport=udp
    15:25:53.656 [CM503025]: Call(62): Calling PSTNline:9922651411@(Ln.10010@PSTNGateway01)@[Dev:sip:10010@192.168.0.180:5060;transport=udp]
    15:25:53.610 [CM503004]: Call(62): Route 2: PSTNline:9912651411@(Ln.10010@PSTNGateway01)@[Dev:sip:10010@192.168.0.180:5060;transport=udp,Dev:sip:10020@192.168.0.180:5062;transport=udp]
    15:25:53.610 [CM503004]: Call(62): Route 1: PSTNline:9922651411@(Ln.10010@PSTNGateway01)@[Dev:sip:10010@192.168.0.180:5060;transport=udp,Dev:sip:10020@192.168.0.180:5062;transport=udp]
    15:25:53.594 [CM503010]: Making route(s) to <sip:2651411@192.168.0.90:5060>
    15:25:53.594 [CM505001]: Ext.400: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 6.0.20943.0] PBX contact: [sip:400@127.0.0.1:5060]
    15:25:53.594 [CM503001]: Call(62): Incoming call from Ext.400 to <sip:2651411@192.168.0.90:5060>
    15:25:40.251 [CM503008]: Call(61): Call is terminated
     
  12. lneblett

    lneblett Well-Known Member

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    OK, let's try this as there are far too many issues being raised and very little detail. This is a GXW4104 which is essentially the same as a 4108. In this scenario, I am using two lines and can make calls into the device using the same line and the GXW picks up fine and passes to 3CX. I have a ring group established and both lines are directed there. I have rollover (call busy forward) implemented as well at the carrier. While ringing into the ring group on the first line, the second call comes in and AT&T rolls it to the second line whereupon, because the ring group is busy, I have the call directed to voice mail. I can also call into both lines at the same time (using each line's individual number) and the device passes the calls to 3CX and each call is handled correctly. I have attached some screen shots.
    The second line 10001 is configured the same as the first. I will post more with regard to the GXW set-up shortly. Start with the bottom picture and work your way up.
     

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  13. lneblett

    lneblett Well-Known Member

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    Here are the GXW settings.HOpefully these will help. As far as caller-id, the carrier has this set.
     

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  14. RLester

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    The max simultaneous call is per line right? My setup looks just like you’re but for simultaneous call is set for 1 for each line. I guess your other screenshots did not show up well. How many call can you have with you setup? Is it 4?
     
  15. lneblett

    lneblett Well-Known Member

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    2 is what I have set for all (device and lines). You previously indicated that you could not find the setting on the PsTN Edit Gateway tab, so look at the first post of screen shots and you will see it. I only have two lines into this particular install, so only have 2 lines in the GXW set.
     
  16. lneblett

    lneblett Well-Known Member

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    I am using 2 for calls on both ports and the device itself (you indicated earlier that you did not see the setting for the device. Please look at the screen shot I posted for the edit PSTN gateway...it is there). The next screen shot shows a simultaneous outbound and inbound call. I used the 3CX Windows softphone to make an outbound call and then used my landline to make an inbound call. The GXW and 3CX handles both as you can see. I then used the 3CX softphone and made one outbound to my landline and then placed the call on hold. I then used the softphone again and used line 2 (on the softphone) to place a second outbound call to my office phone. The simultaneous calls went hrough fine. 3CX selected the 2nd port to make the first call and then the 1st port for the second call. I never selected which line, I merely dialed and I only have one outbound rule - send to PSTN gateway.
     

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  17. RLester

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    Why do you have the setting for max calls set at 2?
     
  18. lneblett

    lneblett Well-Known Member

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    With only 2 PSTN lines connected, can't do anymore than than that. Keep in mind that this limitation is only for the Gateway aspect and had nothing to do with how many I can do with 3CX itself.
     
  19. RLester

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    My point is the setting is for each account on the gateway and each account are for 1 line on the gateway that can only have 1 call. So why would you set it at 2 calls for each account? ((2+2)=4)

    Just to keep it simple!!
    I have it set to just 1 call on each account and I can make 2 calls in and answer both at the same time. I can even make 2 calls out 1 on each line 1 & 2 at the same time. What my trouble is, call out on line 1, put it on hold, and then call in on line 2 the phone that I am dialing in on just rings and I do NOT here any ring internal.
     
  20. lneblett

    lneblett Well-Known Member

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    Please explain how that is different than the scenario that I did where I made a call out on the softphone and then made a call in and showed you the results? I even showed the port screen from 3CX showing both in and out calls. The only difference is that I called back in using line one which due to it already being busy, the carrier rolled to line 2. Tyr the 2 calls and see if any change.
     
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